Mic covering detection in personal audio devices

ABSTRACT

A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

This U.S. patent application is a Continuation of U.S. patentapplication Ser. No. 13/249,711 filed on Sep. 30, 2011, and claimspriority thereto under 35 U.S.C. § 120. U.S. patent application Ser. No.13/249,711 claims priority under 35 U.S.C. § 119(e) to U.S. ProvisionalPatent Application Ser. No. 61/493,162 filed on Jun. 3, 2011.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates generally to personal audio devices suchas wireless telephones that include noise cancellation, and morespecifically, to a personal audio device in which obstruction of one ofthe microphones used for noise cancellation is detected.

2. Background of the Invention

Wireless telephones, such as mobile/cellular telephones, cordlesstelephones, and other consumer audio devices, such as mp3 players, arein widespread use. Performance of such devices with respect tointelligibility can be improved by providing noise canceling using amicrophone to measure ambient acoustic events and then using signalprocessing to insert an anti-noise signal into the output of the deviceto cancel the ambient acoustic events.

Since the acoustic environment around personal audio devices such aswireless telephones can change dramatically, depending on the sources ofnoise that are present and the position of the device itself, it isdesirable to adapt the noise canceling to take into account suchenvironmental changes. However, adaptive noise canceling circuits can becomplex, consume additional power and can generate undesirable resultsunder certain circumstances.

Therefore, it would be desirable to provide a personal audio device,including a wireless telephone, that provides noise cancellation in avariable acoustic environment.

SUMMARY OF THE INVENTION

The above stated objective of providing a personal audio deviceproviding noise cancellation in a variable acoustic environment, isaccomplished in a personal audio device, a method of operation, and anintegrated circuit.

The personal audio device includes a housing, with a transducer mountedon the housing for reproducing an audio signal that includes both sourceaudio for playback to a listener and an anti-noise signal for counteringthe effects of ambient audio sounds in an acoustic output of thetransducer. A reference microphone is mounted on the housing to providea reference microphone signal indicative of the ambient audio sounds.The personal audio device further includes an adaptive noise-canceling(ANC) processing circuit within the housing for adaptively generating ananti-noise signal from the reference microphone signal such that theanti-noise signal causes substantial cancellation of the ambient audiosounds. An error microphone can also be included for correcting for theelectro-acoustic path from the output of the processing circuit throughthe transducer. The ANC processing circuit monitors the content of theambient audio received from the reference microphone and/or the errormicrophone, and/or the output of a microphone provided for capturingnear-end speech if the personal audio device is a wireless telephone. Bycomparing the audio received from two different microphones, the ANCprocessing circuit can determine if one of the noise-cancelingmicrophones is covered and take action to prevent the anti-noise signalfrom adapting incorrectly or generating an undesirable output.

The foregoing and other objectives, features, and advantages of theinvention will be apparent from the following, more particular,description of the preferred embodiment of the invention, as illustratedin the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is an illustration of a wireless telephone 10 in accordance withan embodiment of the present invention.

FIG. 2 is a block diagram of circuits within wireless telephone 10 inaccordance with an embodiment of the present invention.

FIG. 3 is a block diagram depicting signal processing circuits andfunctional blocks within ANC circuit 30 of CODEC integrated circuit 20of FIG. 2 in accordance with an embodiment of the present invention.

FIG. 4 is a block diagram illustrating functional blocks associated withmic covering operations in the circuit of FIG. 3 in accordance with anembodiment of the present invention.

FIG. 5 is a flowchart of a method of determining that a microphone hasbeen obstructed, in accordance with an embodiment of the presentinvention.

FIG. 6 is a block diagram depicting signal processing circuits andfunctional blocks within an integrated circuit in accordance with anembodiment of the present invention.

DESCRIPTION OF ILLUSTRATIVE EMBODIMENT

The present invention encompasses noise canceling techniques andcircuits that can be implemented in a personal audio device, such as awireless telephone. The personal audio device includes an adaptive noisecanceling (ANC) circuit that measures the ambient acoustic environmentand generates a signal that is injected in the speaker (or othertransducer) output to cancel ambient acoustic events. A referencemicrophone is provided to measure the ambient acoustic environment andan error microphone may be included to provide estimation of anelectro-acoustical path from the output of the ANC circuit through thespeaker. The ANC circuit monitors the content of at least two of thereference microphone signal, the error microphone signal and a speechmicrophone signal provided for capturing near-end speech, in order todetermine whether one of the reference microphone or the errormicrophone are obstructed, e.g., covered with a finger or otherobstruction.

Referring now to FIG. 1, a wireless telephone 10 is illustrated inaccordance with an embodiment of the present invention is shown inproximity to a human ear 5. Illustrated wireless telephone 10 is anexample of a device in which techniques in accordance with embodimentsof the invention may be employed, but it is understood that not all ofthe elements or configurations embodied in illustrated wirelesstelephone 10, or in the circuits depicted in subsequent illustrations,are required in order to practice the invention recited in the Claims.Wireless telephone 10 includes a transducer such as speaker SPKR thatreproduces distant speech received by wireless telephone 10, along withother local audio event such as ringtones, stored audio programmaterial, injection of near-end speech (i.e., the speech of the user ofwireless telephone 10) to provide a balanced conversational perception,and other audio that requires reproduction by wireless telephone 10,such as sources from web-pages or other network communications receivedby wireless telephone 10 and audio indications such as battery low andother system event notifications. A near-speech microphone NS isprovided to capture near-end speech, which is transmitted from wirelesstelephone 10 to the other conversation participant(s).

Wireless telephone 10 includes adaptive noise canceling (ANC) circuitsand features that inject an anti-noise signal into speaker SPKR toimprove intelligibility of the distant speech and other audio reproducedby speaker SPKR. A reference microphone R is provided for measuring theambient acoustic environment, and is positioned away from the typicalposition of a user's mouth, so that the near-end speech is minimized inthe signal produced by reference microphone R. A third microphone, errormicrophone E is provided in order to further improve the ANC operationby providing a measure of the ambient audio combined with the audioreproduced by speaker SPKR close to ear 5, when wireless telephone 10 isin close proximity to ear 5. Exemplary circuit 14 within wirelesstelephone 10 include an audio CODEC integrated circuit 20 that receivesthe signals from reference microphone R, near speech microphone NS anderror microphone E and interfaces with other integrated circuits such asan RF integrated circuit 12 containing the wireless telephonetransceiver. In other embodiments of the invention, the circuits andtechniques disclosed herein may be incorporated in a single integratedcircuit that contains control circuits and other functionality forimplementing the entirety of the personal audio device, such as an MP3player-on-a-chip integrated circuit.

In general, the ANC techniques of the present invention measure ambientacoustic events (as opposed to the output of speaker SPKR and/or thenear-end speech) impinging on reference microphone R, and by alsomeasuring the same ambient acoustic events impinging on error microphoneE, the ANC processing circuits of illustrated wireless telephone 10adapt an anti-noise signal generated from the output of referencemicrophone R to have a characteristic that minimizes the amplitude ofthe ambient acoustic events at error microphone E. Since acoustic pathP(z) extends from reference microphone R to error microphone E, the ANCcircuits are essentially estimating acoustic path P(z) combined withremoving effects of an electro-acoustic path S(z) that represents theresponse of the audio output circuits of CODEC IC 20 and theacoustic/electric transfer function of speaker SPKR including thecoupling between speaker SPKR and error microphone E in the particularacoustic environment, which is affected by the proximity and structureof ear 5 and other physical objects and human head structures that maybe in proximity to wireless telephone 10, when wireless telephone is notfirmly pressed to ear 5. While the illustrated wireless telephone 10includes a two microphone ANC system with a third near speech microphoneNS, some aspects of the present invention may be practiced in a systemthat does not include separate error and reference microphones, or awireless telephone uses near speech microphone NS to perform thefunction of the reference microphone R. Also, in personal audio devicesdesigned only for audio playback, near speech microphone NS willgenerally not be included, and the near-speech signal paths in thecircuits described in further detail below can be omitted, withoutchanging the scope of the invention, other than to limit the optionsprovided for input to the microphone covering detection schemes.

Referring now to FIG. 2, circuits within wireless telephone 10 are shownin a block diagram. CODEC integrated circuit 20 includes ananalog-to-digital converter (ADC) 21A for receiving the referencemicrophone signal and generating a digital representation ref of thereference microphone signal, an ADC 21B for receiving the errormicrophone signal and generating a digital representation err of theerror microphone signal, and an ADC 21C for receiving the near speechmicrophone signal and generating a digital representation ns of theerror microphone signal. CODEC IC 20 generates an output for drivingspeaker SPKR from an amplifier A1, which amplifies the output of adigital-to-analog converter (DAC) 23 that receives the output of acombiner 26. Combiner 26 combines audio signals from internal audiosources 24, the anti-noise signal generated by ANC circuit 30, which byconvention has the same polarity as the noise in reference microphonesignal ref and is therefore subtracted by combiner 26, a portion of nearspeech signal ns so that the user of wireless telephone 10 hears theirown voice in proper relation to downlink speech ds, which is receivedfrom radio frequency (RF) integrated circuit 22 and is also combined bycombiner 26. Near speech signal is also provided to RF integratedcircuit 22 and is transmitted as uplink speech to the service providervia antenna ANT.

Referring now to FIG. 3, details of ANC circuit 30 are shown inaccordance with an embodiment of the present invention. Adaptive filter32 receives reference microphone signal ref and under idealcircumstances, adapts its transfer function W(z) to be P(z)/S(z) togenerate the anti-noise signal. The coefficients of adaptive filter 32are controlled by a W coefficient control block 31 that uses acorrelation of two signals to determine the response of adaptive filter32, which generally minimizes the error, in a least-mean squares sense,between those components of reference microphone signal ref and errormicrophone signal err. The signals compared by W coefficient controlblock 31 are the reference microphone signal ref as shaped by a copy ofan estimate of the response of path S(z) provided by filter 34B andanother signal that includes error microphone signal err. Bytransforming reference microphone signal ref with a copy of the estimateof the response of path S(z), SE_(COPY)(z), and minimizing thedifference between the resultant signal and error microphone signal err,adaptive filter 32 adapts to the desired response of P(z)/S(z). Inaddition to error microphone signal err the signal compared to theoutput of filter 34B by W coefficient control block 31 includes aninverted amount of downlink audio signal ds that has been processed byfilter response SE(z), of which response SE_(COPY)(z) is a copy. Byinjecting an inverted amount of downlink audio signal ds adaptive filter32 is prevented from adapting to the relatively large amount of downlinkaudio present in error microphone signal err and by transforming thatinverted copy of downlink audio signal ds with the estimate of theresponse of path S(z), the downlink audio that is removed from errormicrophone signal err before comparison should match the expectedversion of downlink audio signal ds reproduced at error microphonesignal err, since the electrical and acoustical path of S(z) is the pathtaken by downlink audio signal ds to arrive at error microphone E.

To implement the above, adaptive filter 34A has coefficients controlledby SE coefficient control block 33, which compares downlink audio signalds and error microphone signal err after removal of the above-describedfiltered downlink audio signal ds, that has been filtered by adaptivefilter 34A to represent the expected downlink audio delivered to errormicrophone E, and which is removed from the output of adaptive filter34A by a combiner 36. SE coefficient control block 33 correlates theactual downlink speech signal ds with the components of downlink audiosignal ds that are present in error microphone signal err. Adaptivefilter 34A is thereby adapted to generate a signal from downlink audiosignal ds, that when subtracted from error microphone signal err,contains the content of error microphone signal err that is not due todownlink audio signal ds. Event detection and control logic 38 performvarious actions in response to various events in conformity with variousembodiments of the invention, as will be disclosed in further detailbelow.

Since adaptive filter 32 generates the anti-noise signal from referencemicrophone signal ref, if reference microphone R is covered by a fingeror other obstruction, W coefficient control 31 will either have no inputto drive its adaptation from reference microphone signal ref, or theinput will be sounds made by the movement of the obstruction acrossreference microphone R. The covering of reference microphone R may alsocause reference microphone signal to primarily reflect the output ofspeaker SPKR due to internal coupling, which is very undesirable, as theanti-noise signal would, under those conditions, generally attempt tocancel downlink speech signal ds. In any of the above circumstances, Wcannot properly be adapted without a proper reference microphone signalref and may generate an anti-noise signal that is undesirable. If errormicrophone E is covered by an obstruction, such as a portion oflistener's ear 5, then SE coefficient control 33 will adapt incorrectly,which will also cause W coefficient control 31 to also adaptincorrectly.

Referring now to FIG. 4, details of a technique for detecting microphoneobstruction are shown in accordance with an embodiment of the presentinvention as a block diagram of functional blocks, which may beimplemented as an algorithm by a processor that implements errordetection and control block, but at least a portion of which couldalternatively be implemented in dedicated circuits. Each of themicrophone signals, reference microphone signal ref, error microphonesignal err and near-end speech microphone signal ns, are provided as aninput to a corresponding low-pass filter 62A, 62B or 62C, respectively,which remove components of the corresponding microphone signals havingfrequencies above a cut-off frequency, which may be predetermined, e.g.100 Hz, or which may be adapted to ambient conditions. In any case, thecut-off frequency should generally be below a frequency at whichmulti-path phase differences and reflections may generate amplitudedifferences in the microphone signals, and at which the directivity ofthe microphones may come into play. The outputs of low-pass filters 62A,62B and 62C are provided to corresponding signal level detectors 64A,64B and 64C, respectively, which provide signals indicative of theamplitude of low-frequency components in each of the microphone signals:level signal L_(ref) indicative of the amplitude of low-frequencycomponents in reference microphone signal ref, level signal L_(err)indicative of the amplitude of low-frequency components in errormicrophone signal err, and level signal L_(ns) indicative of theamplitude of low-frequency components in near-end speech microphonesignal ns. Level signals L_(ref), L_(err) and L_(ns) are provided to alevel comparison block 66, which generates control output signals thatsignal the ANC circuits described above to mute the ANC action, i.e.,turn off the anti-noise signal, freeze adapting of W(z) and/or SE(z) andreset the coefficients of W(z) and/or SE(z), depending on the particulardetected conditions.

Referring now to FIG. 5, details of a technique for detecting microphoneobstruction are shown in accordance with an embodiment of the presentinvention in a flowchart. If level signal L_(ns) is larger than levelsignal L_(ref) by a predetermined threshold (decision 70), thenreference microphone R is assumed to be obstructed and action is takento mute the ANC system and freeze the adaptation of W(z) (step 72). Iflevel signal L_(ns) is larger than level signal L_(err) by apredetermined threshold (decision 74), then error microphone E isassumed to be obstructed and action is taken to freeze the adaptation ofW(z) and SE(z), as well as resetting the coefficients of both W(z) andSE(z) to predetermined values (step 76). Until ANC operation isterminated (decision 78), e.g., the wireless telephone is turned off,steps 70-78 are repeated. The flowchart of FIG. 5 is only one example ofa detection methodology that may be employed to determine whethermicrophones are obstructed. For example, in devices without a speechmicrophone ns, the low frequency component of reference microphonesignal ref and error microphone signal err could be compared and actiontaken if the corresponding low frequency level signal L_(ref) or L_(err)exceeded the other by a predetermined amount, indicating that the othermicrophone is covered. Further, the actions taken may be different,e.g., mute ANC alone, or reset all adaptive filters and mute ANC underany covering condition, etc., without deviating from the spirit andscope of the invention.

Referring now to FIG. 6, a block diagram of an ANC system in accordancewith an embodiment of the invention is shown, as may be implementedwithin CODEC integrated circuit 20. Reference microphone signal ref isgenerated by a delta-sigma ADC 41A that operates at 64 timesoversampling and the output of which is decimated by a factor of two bya decimator 42A to yield a 32 times oversampled signal. A delta-sigmashaper 43A spreads the energy of images outside of bands in which aresultant response of a parallel pair of adaptive filter stages 44A and44B will have significant response. Filter stage 44B has a fixedresponse W_(FIXED)(z) that is generally predetermined to provide astarting point at the estimate of P(z)/S(z) for the particular design ofwireless telephone 10 for a typical user. An adaptive portionW_(ADAPT)(z) of the response of the estimate of P(z)/S(z) is provided byadaptive filter stage 44A, which is controlled by a leakyleast-means-squared (LMS) coefficient controller 54A. Leaky LMScoefficient controller 54A is leaky in that the response normalizes toflat or otherwise predetermined response over time when no error inputis provided to cause leaky LMS coefficient controller 54A to adapt.Providing a leaky controller prevents long-term instabilities that mightarise under certain environmental conditions, and in general makes thesystem more robust against particular sensitivities of the ANC response.

As in the example of FIG. 3, in the system depicted in FIG. 6, thereference microphone signal is filtered by a copy SE_(COPY)(z) of theestimate of S(z), by a filter 51 that has a response SE_(COPY)(z), theoutput of which is decimated by a factor of 32 by a decimator 52A toyield a baseband audio signal that is provided, through an infiniteimpulse response (IIR) filter 53A to leaky LMS 54A. The error microphonesignal err is generated by a delta-sigma ADC 41C that operates at 64times oversampling and the output of which is decimated by a factor oftwo by a decimator 42B to yield a 32 times oversampled signal. As in thesystem of FIG. 3, an amount of downlink audio ds that has been filteredby an adaptive filter to apply response S(z) is removed from errormicrophone signal err by a combiner 46C, the output of which isdecimated by a factor of 32 by a decimator 52C to yield a baseband audiosignal that is provided, through an infinite impulse response (IIR)filter 53B to leaky LMS 54A. Response S(z) is produced by anotherparallel set of adaptive filter stages 55A and 55B, one of which, filterstage 55B has fixed response SE_(FIXED)(z), and the other of which,filter stage 55A has an adaptive response SE_(ADAPT)(z) controlled byleaky LMS coefficient controller 54B. The outputs of adaptive filterstages 55A and 55B are combined by a combiner 46E. Similar to theimplementation of filter response W(z) described above, responseSE_(FIXED)(z) is generally a predetermined response known to provide asuitable starting point under various operating conditions forelectrical/acoustical path S(z). A separate control value is provided inthe system of FIG. 6 to control adaptive filter 51 that has a responseSE_(COPY)(z), and which is shown as a single adaptive filter stage.However, adaptive filter 51 could alternatively be implemented using twoparallel stages and the same control value used to control adaptivefilter stage 55A could then be used to control the adaptive stage in theimplementation of adaptive filter 51. The inputs to leaky LMS controlblock 54B are also at baseband, provided by decimating downlink audiosignal ds by a decimator 52B that decimates by a factor of 32 after acombiner 46C has removed the signal generated from the combined outputsof adaptive filter stage 55A and filter stage 55B that are combined byanother combiner 46E. The output of combiner 46C represents errormicrophone signal err with the components due to downlink audio signalds removed, which is provided to LMS control block MB after decimationby decimator 52B. The other input to LMS control block 54B is thebaseband signal produced by decimator 52C.

The above arrangement of baseband and oversampled signaling provides forsimplified control and reduced power consumed in the adaptive controlblocks, such as leaky LMS controllers 54A and 54B, while providing thetap flexibility afforded by implementing adaptive filter stages 44A-44B,55A-55B and adaptive filter 51 at the oversampled rates. The remainderof the system of FIG. 6 includes a combiner 46D that combines downlinkaudio ds with internal audio is and a portion of near-end speech thathas been generated by sigma-delta ADC 41B and filtered by a sidetoneattenuator 56 to prevent feedback conditions. The output of combiner 46Dis shaped by a sigma-delta shaper 43B that provides inputs to filterstages 55A and 55B that has been shaped to shift images outside of bandswhere filter stages 55A and 55B will have significant response.

In accordance with an embodiment of the invention, the output ofcombiner 46D is also combined with the output of adaptive filter stages44A-44B that have been processed by a control chain that includes acorresponding hard mute block 45A, 45B for each of the filter stages, acombiner 46A that combines the outputs of hard mute blocks 45A, 45B, asoft mute 47 and then a soft limiter 48 to produce the anti-noise signalthat is subtracted by a combiner 46B with the source audio output ofcombiner 46D. The output of combiner 46B is interpolated up by a factorof two by an interpolator 49 and then reproduced by a sigma-delta DAC 50operated at the 64× oversampling rate. The output of DAC 50 is providedto amplifier A1, which generates the signal delivered to speaker SPKR.

Event detection and control block 38 receives various inputs for eventdetection, such as the output of decimator 52C, which represents howwell the ANC system is canceling acoustic noise as measured at errormicrophone E, the output of decimator 52A, which represents the ambientacoustic environment shaped by path SE(z), downlink audio signal ds, andnear-end speech signal ns. Event detection and control block 38 alsoreceives error microphone signal err, after removal of the components oferror microphone signal due to downlink audio signal ds, and alsoreceives reference microphone signal ref. Event detection and controlblock 38 also includes circuits and/or processing algorithmsimplementing the above-described microphone covering detection and ANCcontrol techniques. Depending on detected acoustic events, or otherenvironmental factors such as the position of wireless telephone 10relative to ear 5 event detection and control block 38 will generate thecontrol outputs described above, along with various other outputs, whichare not shown in FIG. 6 for clarity, but that may control, among otherelements, whether hard mute blocks 45A-45B are applied, characteristicsof mute 47 and limiter 48, whether leaky LMS control blocks 54A and 54Bare frozen or reset, and in some embodiments of the invention, whatfixed responses are selected for the fixed portion of the adaptivefilters, e.g., adaptive filter stages 44B and 55B.

Each or some of the elements in the system of FIG. 6, as well in as theexemplary circuits of FIGS. 2-4, can be implemented directly in logic,or by a processor such as a digital signal processing (DSP) coreexecuting program instructions that perform operations such as theadaptive filtering and LMS coefficient computations. While the DAC andADC stages are generally implemented with dedicated mixed-signalcircuits, the architecture of the ANC system of the present inventionwill generally lend itself to a hybrid approach in which logic may be,for example, used in the highly oversampled sections of the design,while program code or microcode-driven processing elements are chosenfor the more complex, but lower rate operations such as computing thetaps for the adaptive filters and/or responding to detected events suchas those described herein.

While the invention has been particularly shown and described withreference to the preferred embodiments thereof, it will be understood bythose skilled in the art that the foregoing and other changes in form,and details may be made therein without departing from the spirit andscope of the invention.

What is claimed is:
 1. A personal audio device, comprising: a personalaudio device housing; a transducer mounted on the housing forreproducing an audio signal including both source audio for playback toa listener and an anti-noise signal for countering the effects ofambient audio sounds in the proximity of an acoustic output of thetransducer; a plurality of microphones, including a first microphonemounted on the housing that, when unobstructed, provides a firstmicrophone signal indicative of the ambient audio sounds, wherein asecond microphone of the plurality of microphones is mounted on thehousing and that, when unobstructed, provides a second microphone signalindicative of the ambient audio sounds; and a processing circuit thatimplements a first adaptive filter having a response that generates theanti-noise signal from the first microphone signal, a second adaptivefilter for generating shaped source audio from the source audio and acombiner for removing the shaped source audio from the second microphonesignal to generate an error signal provided to a coefficient controlblock that controls coefficients of the first adaptive filter, whereinthe processing circuit implements a first signal level detector fordetecting a first amplitude of a given one of the first microphonesignal or the second microphone signal to generate a microphone levelsignal and a second signal level detector for detecting a secondamplitude of a microphone signal provided by one of the plurality ofmicrophones other than the microphone providing the given microphonesignal to generate a reference level signal, wherein the processingcircuit compares the microphone level signal and the reference levelsignal, in response to determining that differences between themicrophone level signal and the reference level signal indicate that thefirst microphone is at least partially obstructed, resets adaptation ofa given one of at least one of the first adaptive filter or the secondadaptive filter by setting coefficients of the given filter to apredetermined fixed value to prevent the anti-noise signal from beinggenerated erroneously.
 2. The personal audio device of claim 1, whereinthe first signal level detector detects an amplitude of the referencemicrophone signal.
 3. The personal audio device of claim 2, wherein thesecond microphone is an error microphone that provides an errormicrophone signal indicative of the acoustic output of the transducer,wherein the second signal level detector detects the second amplitude ofthe error microphone signal.
 4. The personal audio device of claim 3,wherein the plurality of microphones further includes a speechmicrophone provided for capturing near end speech of a user of thepersonal audio device and providing a speech signal indicative of thenear end speech, and wherein the processing circuit halts the adaptationof the given adaptive filter in response to determining that differencesbetween the microphone level signal, the reference level signal, andanother reference level generated by detecting an amplitude of thespeech signal indicate that the reference microphone or the errormicrophone is at least partially obstructed.
 5. The personal audiodevice of claim 1, wherein the processing circuit further, in responseto determining that the differences between the microphone level signaland the reference level signal indicate that the first microphone is atleast partially obstructed, mutes the anti-noise signal.
 6. The personalaudio device of claim 1, wherein the processing circuit further filtersthe first microphone signal and the second microphone signal to retainonly frequencies below a cut-off frequency at inputs to the first signallevel detector and the second signal level detector in order todetermine that the given microphone is at least partially obstructed. 7.The personal audio device of claim 6, wherein the cutoff frequency issubstantially equal to 100 Hz.
 8. The personal audio device of claim 1,wherein the given microphone signal is the first microphone signal andthe microphone level signal is a first microphone level signal so thatthe first signal level detector detects the first amplitude of the firstmicrophone signal to generate the first microphone level signal, whereinthe processing circuit further implements a third signal level detectorfor detecting the amplitude of the second microphone signal to generatea second microphone level signal, wherein the second signal leveldetector detects the second amplitude of a third microphone signalprovided by another one of the plurality of microphones other than thefirst microphone and the second microphone to generate the referencelevel signal, wherein the processing circuit compares the firstmicrophone level signal and the reference level signal, and in responseto determining that differences between the first microphone levelsignal and the reference level signal indicate that the first microphoneis at least partially obstructed, resets adaptation of the given one ofat least one of the first adaptive filter or the second adaptive filterby setting coefficients of the given filter to the predetermined fixedvalue to prevent the anti-noise signal from being generated erroneously,and wherein the processing circuit compares the second microphone levelsignal and the reference level signal and in response to determiningthat second differences between the second microphone level signal andthe reference level signal indicate that the second microphone is atleast partially obstructed, halts adaptation of at least one of thefirst adaptive filter and the second adaptive filter.
 9. A method ofpreventing production of erroneous anti-noise in a personal audio devicehaving adaptive noise canceling, the method comprising: producing anacoustic output with a transducer, the acoustic output including bothsource audio for playback to a listener and an anti-noise signal forcountering the effects of ambient audio sounds in the proximity of anacoustic output of the transducer; first measuring the ambient audiosounds with a first microphone of a plurality of microphones to generatea first microphone signal; generating the anti-noise signal from thefirst microphone signal with a first adaptive filter; second measuringthe ambient audio sounds with second microphone of the plurality ofmicrophones to generate a second microphone signal; first detecting afirst amplitude of a given one of the first microphone signal or thesecond microphone signal to generate a microphone level signal; seconddetecting a second amplitude of one of the plurality of microphonesother than the given microphone signal to generate a reference levelsignal; comparing the microphone level signal and a signal level of oneof the plurality of microphones other than the first microphone todetermine differences between the microphone level signal and thereference level signal; determining whether the first microphone is atleast partially obstructed from a result of the comparing; in responseto determining that the first microphone is at least partiallyobstructed, resetting adaptation of a given one of at least one of thefirst adaptive filter or the second adaptive filter by settingcoefficients of the given adaptive filter to predetermined fixed valuesto prevent the anti-noise signal from being generated erroneously. 10.The method of claim 9, wherein the first detecting detects a level ofthe reference microphone signal.
 11. The method of claim 10, wherein thesecond microphone is an error microphone that provides an errormicrophone signal indicative of the acoustic output of the transducer,wherein the second detecting detects the second amplitude of the errormicrophone signal.
 12. The method of claim 11, wherein the plurality ofmicrophones includes a speech microphone provided for capturing near endspeech of a user of the personal audio device and providing a speechsignal indicative of the near end speech, wherein the comparingdetermines that differences between the microphone level signal, thereference level signal, and another reference level generated bydetecting an amplitude of the speech signal indicate that the referencemicrophone or the error microphone is at least partially obstructed, andwherein the method further comprises in response to determining that thereference microphone or the error microphone is at least partiallyobstructed, halting adaptation of the given adaptive filter.
 13. Themethod of claim 9, further comprising muting the anti-noise signal inresponse to determining that the differences between the microphonelevel signal and the reference level signal indicate that the firstmicrophone is at least partially obstructed.
 14. The method of claim 9,wherein the comparing detects the differences between the firstmicrophone signal and the second microphone signal only for frequenciesbelow a cut-off frequency in order to determine that the firstmicrophone is at least partially obstructed.
 15. The method of claim 9,wherein the given microphone signal is the first microphone signal andthe microphone level signal is a first microphone level signal so thatthe first detecting detects the first amplitude of the first microphonesignal to generate the first microphone level signal, wherein the seconddetecting detects the second amplitude of a third microphone signalprovided by another one of the plurality of microphones other than thefirst microphone and the second microphone to generate the referencelevel signal, wherein the comparing comprises comparing the firstmicrophone level signal and the reference level signal, and wherein themethod further comprises: third detecting the amplitude of the secondmicrophone signal to generate a second microphone level signal;comparing the second microphone level signal and the reference levelsignal to determined second differences between the second microphonelevel signal and the reference level signal; determining that seconddifferences between the second microphone level signal and the referencelevel signal indicate that the second microphone is at least partiallyobstructed; and responsive to determining that the second microphone isat least partially obstructed, halting adaptation of at least one of thefirst adaptive filter and the second adaptive filter.
 16. An integratedcircuit for implementing at least a portion of a personal audio device,comprising: an output for providing a signal to a transducer includingboth source audio for playback to a listener and an anti-noise signalfor countering the effects of ambient audio sounds in an acoustic outputof the transducer; a first microphone input of a plurality of microphoneinputs for receiving a first microphone signal indicative of the ambientaudio sounds from a first microphone; a second microphone input of theplurality of microphone inputs for receiving a second microphone signalindicative of the ambient audio sounds from a second microphone; and aprocessing circuit that implements a first adaptive filter having aresponse that generates the anti-noise signal from the first microphonesignal, a second adaptive filter for generating shaped source audio fromthe source audio and a combiner for removing the shaped source audiofrom the second microphone signal to generate an error signal providedto a coefficient control block that controls coefficients of the firstadaptive filter, wherein the processing circuit implements a firstsignal level detector for detecting a first amplitude of a given one ofthe first microphone signal or the second microphone signal to generatea microphone level signal and second signal level detector for detectinga second amplitude of a microphone signal provided by one of theplurality of microphone inputs other than the given microphone signal togenerate a reference level signal, wherein the processing circuitcompares the microphone level signal and the reference level signal, inresponse to determining that differences between the microphone levelsignal and the reference level signal indicate that the first microphoneis at least partially obstructed, resets adaptation of a given one of atleast one of the first adaptive filter or the second adaptive filter bysetting coefficients of the given filter to predetermined fixed valuesto prevent the anti-noise signal from being generated erroneously. 17.The integrated circuit of claim 16, wherein the first signal leveldetector detects an amplitude of the reference microphone signal. 18.The integrated circuit of claim 17, wherein the second microphone signalis an error microphone signal indicative of the acoustic output of thetransducer, wherein the second signal level detector detects the secondamplitude of the error microphone signal.
 19. The integrated circuit ofclaim 18, wherein the plurality of microphone inputs includes a speechmicrophone input for receiving a speech signal indicative of near endspeech, and wherein the processing circuit halts the adaptation of thegiven adaptive filter in response to determining that differencesbetween the microphone level signal, the reference level signal, andanother reference level generated by detecting an amplitude of thespeech signal indicate that the first microphone or the secondmicrophone is at least partially obstructed.
 20. The integrated circuitof claim 16, wherein the processing circuit further, in response todetermining that the differences between the microphone level signal andthe reference level signal indicate that the first microphone is atleast partially obstructed, mutes the anti-noise signal.
 21. Theintegrated circuit of claim 16, wherein the processing circuit furtherfilters the first microphone signal and the second microphone signal toretain only frequencies below a cut-off frequency at inputs to the firstsignal level detector and the second signal level detector in order todetermine that the first microphone is at least partially obstructed.22. The integrated circuit of claim 21, wherein the cutoff frequency issubstantially equal to 100 Hz.
 23. The integrated circuit of claim 16,wherein the given microphone signal is the first microphone signal andthe microphone level signal is a first microphone level signal so thatthe first signal level detector detects the first amplitude of the firstmicrophone signal to generate the first microphone level signal, whereinthe processing circuit further implements a third signal level detectorfor detecting the amplitude of the second microphone signal to generatea second microphone level signal, wherein the second signal leveldetector detects the second amplitude of a third microphone signalprovided by another one of the plurality of microphone inputs other thanthe first microphone and the second microphone to generate the referencelevel signal, wherein the processing circuit compares the firstmicrophone level signal and the reference level signal, and in responseto determining that differences between the first microphone levelsignal and the reference level signal indicate that the first microphoneis at least partially obstructed, resets adaptation of the given one ofat least one of the first adaptive filter or the second adaptive filterby setting coefficients of the given filter to the predetermined fixedvalue to prevent the anti-noise signal from being generated erroneously,and wherein the processing circuit compares the second microphone levelsignal and the reference level signal and in response to determiningthat second differences between the second microphone level signal andthe reference level signal indicate that the second microphone is atleast partially obstructed, halts adaptation of at least one of thefirst adaptive filter and the second adaptive filter.